audiointerleave.c 4.4 KB

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  1. /*
  2. * Audio Interleaving functions
  3. *
  4. * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
  5. *
  6. * This file is part of FFmpeg.
  7. *
  8. * FFmpeg is free software; you can redistribute it and/or
  9. * modify it under the terms of the GNU Lesser General Public
  10. * License as published by the Free Software Foundation; either
  11. * version 2.1 of the License, or (at your option) any later version.
  12. *
  13. * FFmpeg is distributed in the hope that it will be useful,
  14. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  16. * Lesser General Public License for more details.
  17. *
  18. * You should have received a copy of the GNU Lesser General Public
  19. * License along with FFmpeg; if not, write to the Free Software
  20. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21. */
  22. #include "libavutil/fifo.h"
  23. #include "avformat.h"
  24. #include "audiointerleave.h"
  25. #include "internal.h"
  26. void ff_audio_interleave_close(AVFormatContext *s)
  27. {
  28. int i;
  29. for (i = 0; i < s->nb_streams; i++) {
  30. AVStream *st = s->streams[i];
  31. AudioInterleaveContext *aic = st->priv_data;
  32. if (st->codec->codec_type == CODEC_TYPE_AUDIO)
  33. av_fifo_free(&aic->fifo);
  34. }
  35. }
  36. int ff_audio_interleave_init(AVFormatContext *s,
  37. const int *samples_per_frame,
  38. AVRational time_base)
  39. {
  40. int i;
  41. if (!samples_per_frame)
  42. return -1;
  43. for (i = 0; i < s->nb_streams; i++) {
  44. AVStream *st = s->streams[i];
  45. AudioInterleaveContext *aic = st->priv_data;
  46. if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
  47. aic->sample_size = (st->codec->channels *
  48. av_get_bits_per_sample(st->codec->codec_id)) / 8;
  49. if (!aic->sample_size) {
  50. av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
  51. return -1;
  52. }
  53. aic->samples_per_frame = samples_per_frame;
  54. aic->samples = aic->samples_per_frame;
  55. aic->time_base = time_base;
  56. aic->fifo_size = 100* *aic->samples;
  57. av_fifo_init(&aic->fifo, 100 * *aic->samples);
  58. }
  59. }
  60. return 0;
  61. }
  62. static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
  63. int stream_index, int flush)
  64. {
  65. AVStream *st = s->streams[stream_index];
  66. AudioInterleaveContext *aic = st->priv_data;
  67. int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
  68. if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
  69. return 0;
  70. av_new_packet(pkt, size);
  71. av_fifo_read(&aic->fifo, pkt->data, size);
  72. pkt->dts = pkt->pts = aic->dts;
  73. pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
  74. pkt->stream_index = stream_index;
  75. aic->dts += pkt->duration;
  76. aic->samples++;
  77. if (!*aic->samples)
  78. aic->samples = aic->samples_per_frame;
  79. return size;
  80. }
  81. int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
  82. int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
  83. int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
  84. {
  85. int i;
  86. if (pkt) {
  87. AVStream *st = s->streams[pkt->stream_index];
  88. AudioInterleaveContext *aic = st->priv_data;
  89. if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
  90. unsigned new_size = av_fifo_size(&aic->fifo) + pkt->size;
  91. if (new_size > aic->fifo_size) {
  92. if (av_fifo_realloc2(&aic->fifo, new_size) < 0)
  93. return -1;
  94. aic->fifo_size = new_size;
  95. }
  96. av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
  97. } else {
  98. // rewrite pts and dts to be decoded time line position
  99. pkt->pts = pkt->dts = aic->dts;
  100. aic->dts += pkt->duration;
  101. ff_interleave_add_packet(s, pkt, compare_ts);
  102. }
  103. pkt = NULL;
  104. }
  105. for (i = 0; i < s->nb_streams; i++) {
  106. AVStream *st = s->streams[i];
  107. if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
  108. AVPacket new_pkt;
  109. while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
  110. ff_interleave_add_packet(s, &new_pkt, compare_ts);
  111. }
  112. }
  113. return get_packet(s, out, pkt, flush);
  114. }